When a speech/audio signal is transmitted using a packet communication system represented by an Internet communication or mobile communication system, a compression/coding technology is often used to enhance transmission efficiency of the speech/audio signal. Furthermore, with regard to multiplexing of signals, the smaller the transmission bit rate of each communication terminal, the more communications can be multiplexed, and therefore for many subscribers to simultaneously communicate, it is desirable to adopt a technique that reduces a transmission bit rate of each communication terminal and enhance the efficiency of channels.
In this respect, there are conventionally disclosed technologies for reducing a transmission bit rate in a communication terminal and base station by acquiring information such as the number of simultaneously accessing users, call loss rate, access waiting time, BER (Bit Error Rate), SIR (Signal Interference Ratio), selecting an appropriate mode from among a plurality of predetermined communication modes according to the information acquired and carrying out communication (e.g., Patent Document 1).
Furthermore, a technique of detecting the presence/absence of speech of a speaker and controlling a transmission bit rate according to its detection result, is also developed. For example, Non-patent Document 1 discloses a technology of detecting the presence/absence of speech of a speaker, transmitting data coded at a high bit rate for a period during which the speaker is speaking (voiced period), coded at a low bit rate for a period during which the speaker is not speaking (unvoiced period) so as to reduce the overall transmission bit rate (e.g., Non-patent Document 1).    Patent Document 1 Japanese Patent Application Laid-Open No. 11-331936    Non-patent Document 1: ANSI/TIA/EIA-96-C, Speech Service Option Standard for Wideband Spread Spectrum Digital Cellular System